You can create as many as 8 accouts. The first account (default account) in the list is used for outgoing calls. You can set any of your accounts to be the default account by press the account for long time and you can see this page:

Tap the 'Setings' icon at the bottom brings you to the 'Settings' page:

SIP Stun and SIP Stun Server: If your aSip is behind a NAT/Firewall and your Sip Provider does not do RTP proxy, you'll likely need a stun server so that your voice traffic can go through the NAT.

CODEC: See answer to question 10.

Outbound Proxy: If you want aSip to make SIP calls through a proxy, set it here. the format is like this: "sip-proxy-domain-or-ip:port". You do not have to specify the ':port' portion if your proxy uses the default sip port 5060.

RTP Port: Tell aSip your local RTP port preference. For example, if setting it to 12000, the local RTP port used by iSip in any SIP call will always be greater than 12000.

VAD: VAD = Voice Activity Detection.RTP Silence Suppression Endpoints sending audio as an RTP stream are not required to send packets during silent periods. The capability to stop sending RTP packets during silent periods is known as "Silence Suppression" or VAD (Voice Activity Detection). Whether to use Silence Suppression is usually a configuration option on endpoints. When processing a stream of RTP packets, here is what RFC 3389 has to say about detecting Silence Suppression: RTP allows discontinuous transmission (silence suppression) on any audio payload format. The receiver can detect silence suppression on the first packet received after the silence by observing that the RTP timestamp is not contiguous with the end of the interval covered by the previous packet even though the RTP sequence number has incremented only by one. The RTP marker bit is also normally set on such a packet.

RFC3605 Support:disable this to compatiable with legancy sip server.The Session Description Protocol (SDP) is used to describe the parameters of media streams used in multimedia sessions. When a session requires multiple ports, SDP assumes that these ports have consecutive numbers. However, when the session crosses a network address translation device that also uses port mapping, the ordering of ports can be destroyed by the translation. To handle this, we propose an extension attribute to SDP.

Prefix for Contacts: you can specify a 'prefix-number' so when you make a call directly from one of your stored contacts, iSip will automatically add the 'prefix-number' to the stored number when dialing

Background Photo:Tap 'Background Photo', you can select image for the background,and you will see the following page:

Most cases you don't need a SIP proxy for your SIP calls. But sometime the network environment you are in may require your SIP calls to go through a SIP proxy. To set it up, you can either add the proxy information through the 'Account Manager' to whichever account that requires a proxy, or you can set it up through the 'Settings' page which applies to all accounts.

They serve the same purpose as explained in the last question. The difference is, the proxy setting in the 'Account Editing' page applies only to that particular account, while the 'outbound proxy' in 'Settings' page applies to all the accounts.
In a word, only when all of your accounts require the same proxy, you need to set the 'outbound proxy' in 'Settings' page. For all the rest scenarios, you should leave it empty.

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